[asterisk-users] Voicemail
Please, I need help. I have problem witch voicemail. -- Starting simple switch on 'Zap/4-1' [Aug 17 21:33:24] NOTICE[11864]: chan_zap.c:7093 ss_thread: Got event 18 (Ring Begin)... [Aug 17 21:33:25] NOTICE[11864]: chan_zap.c:7093 ss_thread: Got event 2 (Ring/Answered)... -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/4-1, SIP/2002,20,tr) in new stack == Using SIP RTP CoS mark 5 -- Called 2002 -- SIP/2002-08238d28 is ringing -- Nobody picked up in 2 ms -- Executing [EMAIL PROTECTED]:3] VoiceMail(Zap/4-1, s) in new stack [Aug 17 21:33:46] WARNING[11864]: app_voicemail.c:3061 leave_voicemail: No entry in voicemail config file for 's' -- Executing [EMAIL PROTECTED]:4] Hangup(Zap/4-1, ) in new stack == Spawn extension (Incoming, s, 4) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail
Miguel Otamendi schrieb:
Please, I need help.
I have problem witch voicemail.
-- Starting simple switch on 'Zap/4-1'
[Aug 17 21:33:24] NOTICE[11864]: chan_zap.c:7093 ss_thread: Got event 18
(Ring Begin)...
[Aug 17 21:33:25] NOTICE[11864]: chan_zap.c:7093 ss_thread: Got event 2
(Ring/Answered)...
-- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack
-- Executing [EMAIL PROTECTED]:2] Dial(Zap/4-1, SIP/2002,20,tr) in
new
stack
== Using SIP RTP CoS mark 5
-- Called 2002
-- SIP/2002-08238d28 is ringing
-- Nobody picked up in 2 ms
-- Executing [EMAIL PROTECTED]:3] VoiceMail(Zap/4-1, s) in new stack
[Aug 17 21:33:46] WARNING[11864]: app_voicemail.c:3061 leave_voicemail: No
entry in voicemail config file for 's'
-- Executing [EMAIL PROTECTED]:4] Hangup(Zap/4-1, ) in new stack
== Spawn extension (Incoming, s, 4) exited non-zero on 'Zap/4-1'
-- Hungup 'Zap/4-1'
Try
VoiceMail(2002);
instead of
VoiceMail(${EXTEN});
As you can see you are at extension s.
Grüße,
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail
Am Dienstag, den 19.08.2008, 02:53 + schrieb Miguel Otamendi: Please, I need help. I have problem witch voicemail. -- Executing [EMAIL PROTECTED]:3] VoiceMail(Zap/4-1, s) in new stack [Aug 17 21:33:46] WARNING[11864]: app_voicemail.c:3061 leave_voicemail: No entry in voicemail config file for 's' -- Executing [EMAIL PROTECTED]:4] Hangup(Zap/4-1, ) in new stack == Spawn extension (Incoming, s, 4) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' Hi Miguel, please see http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail for details about the VoiceMail command. What seems to happen in your setup is that the call runs into the s extension, and then VoiceMail() is called. As you do not specify a voicemail box number, s is taken as a box number, which is probably not what you want. Check extensions.conf and alter the VoiceMail command like VoiceMail(1) instead of VoiceMail(), and define a mailbox number 1 in voicemail.conf (or any number you like, of course). You possibly can also define a mailbox number s in voicemail.conf, but that will run you into trouble if you want to listen to messages from abroad, as s is hard to enter by DTMF touchpad ;-) Not sure if that box s works at all though. The safe bet is to use numeric voicemail box numbers. Regards Anselm smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] seeking hardware recommendation PCI versus PCI Express E1 card (te407p vs te420bf)
Jay R. Ashworth wrote: On Wed, Aug 13, 2008 at 03:06:58PM -0500, Kevin P. Fleming wrote: Keep in mind that even if you use 4 E1 circuits with the card, the total bandwidth consumption of card is approximately 1 megabyte per second (4 times 2 megabits per second), which is drastically below the PCI bus bandwidth of 132 megabytes per second (33 MHz bus with 32-bit transfers). No quad-T1/E1 card will ever be able to saturate a PCI bus, especially not PCI-X or PCI-E. Though, depending on the design of the card and drivers, interrupts/sec may be an issue -- sometimes a deciding issue. Hi, Can you pls elaborate when you have time or give some reading resources about this issue. regds mallah. Cheers, -- jra ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digital-OPSiS is giving you the chance to Win A FREE Astricon Pass
Digital-OPSiS is offering a Free Exhibitors Pass, valued at $695 to one lucky winner. Register at http://www.digital-opsis.com/astricon08 to be part of the Digital-OPSiS team and get all the action of Astricon. For those of you who register we have more offerings * A discount code that entitles you to a 15% discount to all Astricon registrations.* * A free Expo Hall Pass (valued at $50), so you can visit the Astricon Expo and see us at the Digital-OPSiS booth*. The Free Exhibitor Pass includes access to all conference and pre-conference activities. For those who want a bit of background on Asterisk, the Asterisk 123 pre-conference seminar is ideal. Developers won't want to miss the Developer 101 activity, which provides a chance to catch up on the latest additions and changes to the Asterisk code base. Or the Asterisk Ecosystem, where you can learn the latest and greatest of the new products out for Asterisk. The Pass also includes all other conference activities. This Includes * Pre Conference Activities - Tuesday, September 23. Your choice of: * Asterisk 123 * Asterisk Ecosystem * Asterisk Developer 101 * In addition, you will receive access to the following events Tuesday night: * AstriCon Expo Hall Opening Reception * Code Zone Opening Party * All Conference Sessions - Wednesday, Sept. 24 - Thursday, Sept. 25 * All Tutorial Sessions - (Wednesday-Thursday) * All General Session Conference Presentations - (Wednesday-Thursday) * The Asterisk Exhibit Hall - (Tuesday-Thursday) * All Lunches and Breaks * All BOF Sessions * All Panel Discussions The lucky winner will be announced, Friday August 22nd at the Voip Users Conference http://VoipUsersConference.org *Code is valid till September 21st ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] disable auth between two asterisk
Hello list, i wanted to setup a small asterisk+ss7 lab this weekend and just installed asterisk-trunk+ dahdi-complete+libss7+libpri i had only a sangoma A101 card so i used it and 48h after i'm still unable to make the card work in that config. i tried to patch the sangoma drivers thinking that it was just a matter of find_replace(zaptel BY dahdi) but i discovered that i should also do a find_replace(zt_ BY dahdi_) and also find_replace(INSTALL_DIR/kernel BY INSTALL_DIR/kernel/include/drivers/dahdi) after all that the sangoma drivers still fails to install due to some declaration that are in the zaptel.h and not in the file /include/drivers/dahdi/user.h ... THE END of tries i give up and go to bed ! Anyone know about any patch of sangoma drivers that support directly dahdi ? regards, -- -- Your next Partner ! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?
Do you have a firewall enabled? If so, disable it. Else run /etc/init.d/iptables stop (or equivalent) and try again. regards, Drew Jonathan Miller wrote: We put a 3c509 in, just for posterity, and it did not help the issue. I verified that the NIC is not sharing any interrupts, that there is no excessive disk wait, and that asterisk thinks it is sending the packets. They are simply not making it to the physical interface. Is there somewhere else I can look or something else I can do? How would I go about prioritizing the asterisk process? There's not a lot of processes running besides asterisk itself, dhcpd, tftpd and postfix. Really just kernel stuff after that... This is killing me. Voice drops out for various periods (between 1/2 and 5 seconds) and lost packets do not show up with a rtp rtcp stats... This is weird. Any help you can offer would be appreciated. We spent 6 hours on phone with Digium support yesterday and could not locate an issue within asterisk itself. -Jonathan On Wed, Aug 13, 2008 at 9:08 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Wed, Aug 13, 2008 at 7:40 PM, Jonathan Miller [EMAIL PROTECTED] wrote: From what I can determine while troubleshooting a voice-dropping issue, the Asterisk server in my organization has been dropping RTP packets between the asterisk server process and the network interface. I determined this from an RTP debug that showed packets sent to the phone and packets received from the phone during the entire call. A tcpdump done on the server for the interface that would deliver the packet to the wire does not show the packets. Is there somewhere I can look to resolve this? Something anybody has come across? It is happening frequently and with great discomfort to many users. I had upgraded from 1.2.x to latest 1.4.x in attempts to resolve this. I also disabled a lot of COM/LPT and USB devices in the BIOS to free up some IRQ's. no devices are sharing IRQ's at this point, with I thought might have been part of the issue, but has proved to at least not be directly related. These calls are from a PRI to a Cisco 7940 using SIP. There is a Juniper EX switch between the two. Both sides negotiate at 100Mbps/Full Duplex. I have ruled the switch out of the problem as it's not seeing the packets on the wire when the issue is occuring. Please help or point me to someone that can. -Jonathan [EMAIL PROTECTED] Jonathan, It sounds hardware specific to me. Is this a new install or a new problem? If it is a new problem, then what has changed? Is the NIC in question onboard? What hardware are you using? Brands, MoBo, NIC, etc... If I were you, I would remove or disable the NIC and stick a tried and true old school 3Com NIC in the server and try that. Thanks, Steve Totaro -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Aastra 480ci and qualify=yes
I;m using Aastra 480i's 9133i's, 9112i's, and 57i's and none of them have experienced problems with qualify=yes. I;m currently on Asterisk 1.4.17, but I've also tested them with 1.4.14 up to 1.4.19. -- Matt http://www.mattgwatson.ca On Fri, Aug 15, 2008 at 10:59 AM, Drew Gibson [EMAIL PROTECTED] wrote: James Lamanna wrote: Hi, We have a few Aastra 480ci phones and we've noticed that in order to get the phone to receive a call, qualify must be = no. Apparently the Aastras do not respond to the qualify message (or respond in a way Asterisk doesn't understand) and Asterisk thinks the phone is unreachable. However, this now prevents MWI from working properly on the phones. Does anyone know how to get MWI working without qualify? Or how to get qualify working again with the Aastras? We have a number of 480i and one 480ct all setup with qualify=yes (Asterisk 1.2.24) Our inbound call centre seems to be pretty busy and my own MWI lights up far too often. Never had a problem with either. Which version of firmware? Which version of Asterisk? What's in your sip.conf? What error messages show on the console? Anything relevant in the logs? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange putconfig bahviour
Hello, I'm running identical putconfig AMI action against 2 asterisks == Action: updateconfig reload: no SrcFilename: extensions.conf DstFilename: extensions.conf.new Action-00: newcat Cat-00: ami-test-01 Action-01: append Var-01: exten Value-01: 888,1,Noop(888) Cat-01: ami-test-01 Action-02: append Var-02: exten Value-02: 999,1,Noop(999) Cat-02: ami-test-01 == It works ok on one server and returns error on the second: Response: Error Message: Save of config failed I did check for obvious things like permissions in manager.conf Where i should be looking to resolve this issue? Thanks Vadim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange putconfig bahviour
On Mon, 2008-08-18 at 16:03 +, Vadim Lebedev wrote: I'm running identical putconfig AMI action against 2 asterisks snip Value-01: 888,1,Noop(888) It may or may not be related, but those greater-than signs () in there look wrong, and may be causing some problems. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Stops...where to look?
Hi, I am running Asterisk 1.4.21.2 on Fedora Core 2. Looks like The Asterisk Process dies after a few hours...I have full debugging turned on but file /var/log/asterisk/full does not show anything specifc..neither does var/log/messages.. dmesg also shows nothing specific to Asterisk dying/corring...Where does Asterisk dump core files if it cores..Any other pointers on where to look would be helpful... Thx.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] US-based echo test servers?
Hi, I'm running a small Asterisk server in the UK, just for personal use. I've been experimenting with various VoIP providers for international calls to PSTN numbers, particularly to the US (often California). My results, to date, have been very variable indeed, so much so that I'm considering getting a suitable card and using the PSTN. I have found a VoIP provider with an excellent reputation, and it gives very good quality. However, I seem to get quite a bit of delay at times, enough to make conversation awkward. As the setup at the far end was not completely trivial, I'm not 100% sure the problem was in my connection, but I'd like to test that. Are there any US numbers I can call to get an Asterisk-style echo test? Ideally, a California-based numnber, so I can try to call it from an ordinary PSTN phone here, and compare calling via VoIP, and see if there's an appreciable difference in the delay/quality. I don't anticipate using this for very long, so it doesn't necessarily need to be a free service. Failing that, does anyone have access to a US-based Asterisk server which would allow me to make connections to its echo test? Presumably, if I had this, I could rent a PSTN number from a US-based provider, and point it to the appropriate SIP/IAX address. I expect my total usage would be just a few minutes, though having the facility available for a few weeks would be helpful, to allow me to play around with various options. Again, I'd be willing to pay a modest amount for this. Thanks in advance for any suggestions! Best wishes, Nikhil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 audible hold reminder?
On Fri, Aug 15, 2008 at 12:27:16PM -0500, [EMAIL PROTECTED] wrote: Hello, I have recently setup my first PBX and am wondering if there might be a way to send audible notification to the cisco 7960 phone when a call is put on hold. We lost a call due to a customer being on hold and forgotten about (yikes). Is there a way to get the phone to beep or ring down the same or other SIP channels after a certain amount of time on hold? Yes and no. (I am on the SIP version 8.9) In the config file for the phone: call_hold_ringback: 1 This option means that if there is a call on hold, and the handset is replaced (say, after ending another call) then the held call will ring again at the handset. I don't think there is a way (on the handset) to set a held call timeout to re-ring on the phone. If you park the call with asterisk instead of holding it, then the call park option allows calls to come back to the person who parked them after a set timeout. You may be able to do something else in asterisk, though not sure what. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 134-138 Borough High Street, London SE1 1LB Registered in England 3137929 at 3 Park Road, Peterborough, PE1 2UX ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cmdRecord issue related to iax2 received mini frame before first full voice frame?
A more detailed explanation of what the above warning means/implies, and how or why it might be preventing recordings would be greatly appreciated. A mini frame is simply a frame containing minimal information about the call itself (the meta data), and a full frame contains all of the meta-data information. Sending mini-frames is part of the IAX protocol, as a way of saving significant bandwidth over the course of a call. However, a mini frame cannot be interpreted correctly independently of a full frame. In every media stream, a full frame is send approximately once every 60 seconds, to sync the timestamps. I'm running Asterisk 1.4.11 on debian Etch. There have been many changes, bug fixes, and even security issues fixed since 1.4.11. I'd really recommend that you try something more recent (and even the latest, because we fixed 2 security issues in the latest release, 1.4.21.2). Thank you very much for this clear explanation of what is going on. The implication then is that this issue could indeed be having a significant impact on whether calls are eventually recorded? I plan to look into upgrading our distribution, but this appears to be impacting 1/10 calls coming through our system at the moment so I wonder if there is some/any way to help make sure that that first full voice frame gets sent before the miniframes get pushed through? I also wonder if there is a point after which the call is lost, regardless of whether the first full voice frame comes through? could other issues like mismatches between the system clocks on the respective machines also be impacting performance? thanks a bunch! -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange putconfig bahviour
Value-01: 888,1,Noop(888) It may or may not be related, but those greater-than signs () in there look wrong, and may be causing some problems. Without them you can't get exten=888,1,Dial(888) if your remove '' you get exten=888,1,Dial(888) in extensions.conf which is BAD. And anyway as i've said it wokr ok on one of the atsreisk and fails on second one. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk vs c-client issues
Lee Lundrigan wrote: Russell Bryant wrote: Lee Lundrigan wrote: Hi everyone, Are there any incompatibility issues between asterisk and the c-client using SSL? When I enable SSL I get the error: *pbx.c:1832 pbx_extension_helper: No application 'VoiceMailMain' *whenever I am trying to access voicemail. But when SSL is disabled everything works great, just like its supposed to with imap. Any ideas? It looks like app_voicemail is failing to load when you build c-client with SSL support. Try running module load app_voicemail.so from the Asterisk CLI to see what the error message is. This is everything: londo*CLI module load app_voicemail.so [Aug 15 12:45:24] WARNING[14459]: loader.c:363 load_dynamic_module: Error loading module 'app_voicemail.so': /usr/lib/asterisk/modules/app_voicemail.so: cannot restore segment prot after reloc: Permission denied Isn't that an SELinux issue? Use something like this to fix it: # chcon -t texrel_shlib_t /location/of/voicemail.so Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 43
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], altrimenti vi risponderò al mio rientro. Dimitri Osler I will be on vacation until Tuesday 19th of August with limited access to voice and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on my return. Dimitri Osler ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US-based echo test servers?
Hey Nikhil, I have some free time right now and would be willing to set this up for you. Just paypal me $1 for the DID(globalpops fee) and like 50 cents for minutes. Whatever is left over after you're done testing I can refund to you. Regards, Igor H. Nikhil Nair wrote: Hi, I'm running a small Asterisk server in the UK, just for personal use. I've been experimenting with various VoIP providers for international calls to PSTN numbers, particularly to the US (often California). My results, to date, have been very variable indeed, so much so that I'm considering getting a suitable card and using the PSTN. I have found a VoIP provider with an excellent reputation, and it gives very good quality. However, I seem to get quite a bit of delay at times, enough to make conversation awkward. As the setup at the far end was not completely trivial, I'm not 100% sure the problem was in my connection, but I'd like to test that. Are there any US numbers I can call to get an Asterisk-style echo test? Ideally, a California-based numnber, so I can try to call it from an ordinary PSTN phone here, and compare calling via VoIP, and see if there's an appreciable difference in the delay/quality. I don't anticipate using this for very long, so it doesn't necessarily need to be a free service. Failing that, does anyone have access to a US-based Asterisk server which would allow me to make connections to its echo test? Presumably, if I had this, I could rent a PSTN number from a US-based provider, and point it to the appropriate SIP/IAX address. I expect my total usage would be just a few minutes, though having the facility available for a few weeks would be helpful, to allow me to play around with various options. Again, I'd be willing to pay a modest amount for this. Thanks in advance for any suggestions! Best wishes, Nikhil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which is the correct GUI vor Asterisk 1.4 ?
branches/2.0 All you have to do is click 'Add Service Provider'. -bk Klaus Ruebsam wrote: Which is the correct GUI vor Asterisk 1.4 ? Is it http://svn.digium.com/svn/asterisk-gui/branches/1.0 or http://svn.digium.com/svn/asterisk-gui/branches/2.0 ? Is there any kind of documentation for the proper usage of the GUI as I can´t manage to setup a SIP-account for my SIP-provider. What is the proper way to add additional providers to the dropdown list? Any help appreciated. Thanks in advance. Best regards, Klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US-based echo test servers?
On Mon, Aug 18, 2008 at 8:09 PM, emist [EMAIL PROTECTED] wrote: Hey Nikhil, I have some free time right now and would be willing to set this up for you. Just paypal me $1 for the DID(globalpops fee) and like 50 cents for minutes. Whatever is left over after you're done testing I can refund to you. Oh come on, do you give toll free number or what? Btw i just assigned DID to Echo() app, so Nikhil has what he asked for. And for free :p Regards, Atis Regards, Igor H. Nikhil Nair wrote: Hi, I'm running a small Asterisk server in the UK, just for personal use. I've been experimenting with various VoIP providers for international calls to PSTN numbers, particularly to the US (often California). My results, to date, have been very variable indeed, so much so that I'm considering getting a suitable card and using the PSTN. I have found a VoIP provider with an excellent reputation, and it gives very good quality. However, I seem to get quite a bit of delay at times, enough to make conversation awkward. As the setup at the far end was not completely trivial, I'm not 100% sure the problem was in my connection, but I'd like to test that. Are there any US numbers I can call to get an Asterisk-style echo test? Ideally, a California-based numnber, so I can try to call it from an ordinary PSTN phone here, and compare calling via VoIP, and see if there's an appreciable difference in the delay/quality. I don't anticipate using this for very long, so it doesn't necessarily need to be a free service. Failing that, does anyone have access to a US-based Asterisk server which would allow me to make connections to its echo test? Presumably, if I had this, I could rent a PSTN number from a US-based provider, and point it to the appropriate SIP/IAX address. I expect my total usage would be just a few minutes, though having the facility available for a few weeks would be helpful, to allow me to play around with various options. Again, I'd be willing to pay a modest amount for this. Thanks in advance for any suggestions! Best wishes, Nikhil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Stops...where to look?
Jay Ray wrote: Hi, I am running Asterisk 1.4.21.2 on Fedora Core 2. Looks like The Asterisk Process dies after a few hours...I have full debugging turned on but file /var/log/asterisk/full does not show anything specifc..neither does var/log/messages.. dmesg also shows nothing specific to Asterisk dying/corring...Where does Asterisk dump core files if it cores..Any other pointers on where to look would be helpful... Thx.. Asterisk will typically dump core to the directory from which it is run, assuming that it was run with the -g flag at startup. Some of the startup scripts provided for Asterisk, including safe_asterisk, will dump core to /tmp. Also, Fedora Core 2?! Ouch! ;) Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US-based echo test servers?
Another thing you may want to do is try a simple ping test to the far end host. While this may not always be a reliable way to test lag given that the far end maybe just a proxy and your RTP may be terminating to another device, it still should give you a good idea what your lag times are at least on the signaling end of things. You could also do a traceroute to see how many hops your having to jump through as well. You could use a tool like ngrep to actually see the sip signaling and copy out the media gateway from the SDP if you really wanted to, and do a ping on that. I've done extensive work with international voip origination and termination, and typically I haven't had any problems unless it's going over satellite (lag) or there is a problem at the far end (usually pdd or quality issues). If things keep up, I'd even consider running top during a call to see what kind of utilization your local server is at just to make sure something isn't wrong there either. Hope this helps, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 18, 2008, at 10:41 AM, Nikhil Nair wrote: Hi, I'm running a small Asterisk server in the UK, just for personal use. I've been experimenting with various VoIP providers for international calls to PSTN numbers, particularly to the US (often California). My results, to date, have been very variable indeed, so much so that I'm considering getting a suitable card and using the PSTN. I have found a VoIP provider with an excellent reputation, and it gives very good quality. However, I seem to get quite a bit of delay at times, enough to make conversation awkward. As the setup at the far end was not completely trivial, I'm not 100% sure the problem was in my connection, but I'd like to test that. Are there any US numbers I can call to get an Asterisk-style echo test? Ideally, a California-based numnber, so I can try to call it from an ordinary PSTN phone here, and compare calling via VoIP, and see if there's an appreciable difference in the delay/quality. I don't anticipate using this for very long, so it doesn't necessarily need to be a free service. Failing that, does anyone have access to a US-based Asterisk server which would allow me to make connections to its echo test? Presumably, if I had this, I could rent a PSTN number from a US-based provider, and point it to the appropriate SIP/IAX address. I expect my total usage would be just a few minutes, though having the facility available for a few weeks would be helpful, to allow me to play around with various options. Again, I'd be willing to pay a modest amount for this. Thanks in advance for any suggestions! Best wishes, Nikhil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Stops...where to look?
On Mon, Aug 18, 2008 at 1:50 PM, Mark Michelson [EMAIL PROTECTED] wrote: Jay Ray wrote: Hi, I am running Asterisk 1.4.21.2 on Fedora Core 2. Looks like The Asterisk Process dies after a few hours...I have full debugging turned on but file /var/log/asterisk/full does not show anything specifc..neither does var/log/messages.. dmesg also shows nothing specific to Asterisk dying/corring...Where does Asterisk dump core files if it cores..Any other pointers on where to look would be helpful... Thx.. Asterisk will typically dump core to the directory from which it is run, assuming that it was run with the -g flag at startup. Some of the startup scripts provided for Asterisk, including safe_asterisk, will dump core to /tmp. Also, Fedora Core 2?! Ouch! ;) Mark Michelson I have to second the FC2 sentiment, when was that EOLed? Why not load up CentOS 5 or whatever? As Mark referenced and a good suggestion for your situation is to use safe_asterisk to invoke Asterisk. That way you get a core dump by default in /tmp AND Asterisk will pretty much instantly restart itself. Sure your calls will drop but at least Asterisk will be up and running a moment later, giving you more uptime and time to GDB your core dump. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US-based echo test servers?
Nikhil Nair wrote: Hi, I'm running a small Asterisk server in the UK, just for personal use. I've been experimenting with various VoIP providers for international calls to PSTN numbers, particularly to the US (often California). My results, to date, have been very variable indeed, so much so that I'm considering getting a suitable card and using the PSTN. I have found a VoIP provider with an excellent reputation, and it gives very good quality. However, I seem to get quite a bit of delay at times, enough to make conversation awkward. As the setup at the far end was not completely trivial, I'm not 100% sure the problem was in my connection, but I'd like to test that. Are there any US numbers I can call to get an Asterisk-style echo test? Ideally, a California-based numnber, so I can try to call it from an ordinary PSTN phone here, and compare calling via VoIP, and see if there's an appreciable difference in the delay/quality. I don't anticipate using this for very long, so it doesn't necessarily need to be a free service. Failing that, does anyone have access to a US-based Asterisk server which would allow me to make connections to its echo test? Presumably, if I had this, I could rent a PSTN number from a US-based provider, and point it to the appropriate SIP/IAX address. I expect my total usage would be just a few minutes, though having the facility available for a few weeks would be helpful, to allow me to play around with various options. Again, I'd be willing to pay a modest amount for this. Thanks in advance for any suggestions! Best wishes, Nikhil. Nikhil, Can't help out on the California number, but IdeaSIP accepts incoming calls to its echo test. [EMAIL PROTECTED] Alternatively, for a west-coast PSTN number to call, you could get a number from IPKall.com, and forward it to [EMAIL PROTECTED], and have a rough estimate. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US-based echo test servers?
Interesting response there Atis. I'm glad you were able to help Nikhil out for free. I for one don't have free DIDs and anyone can verify that I indeed pay $1 setup fee to globalpops per DID(toll-free or otherwise). All I asked was for _my operational fee_ ($1 + whatever minutes he uses charged at 0.004c) and I was to make no profit out of this for a person I don't even know to begin with to allow them to use my switch. However, with responses like yours I think I'll be more conservative in regards to whom I offer my help next time. Regards, Igor H. Atis Lezdins wrote: On Mon, Aug 18, 2008 at 8:09 PM, emist [EMAIL PROTECTED] wrote: Hey Nikhil, I have some free time right now and would be willing to set this up for you. Just paypal me $1 for the DID(globalpops fee) and like 50 cents for minutes. Whatever is left over after you're done testing I can refund to you. Oh come on, do you give toll free number or what? Btw i just assigned DID to Echo() app, so Nikhil has what he asked for. And for free :p Regards, Atis Regards, Igor H. Nikhil Nair wrote: Hi, I'm running a small Asterisk server in the UK, just for personal use. I've been experimenting with various VoIP providers for international calls to PSTN numbers, particularly to the US (often California). My results, to date, have been very variable indeed, so much so that I'm considering getting a suitable card and using the PSTN. I have found a VoIP provider with an excellent reputation, and it gives very good quality. However, I seem to get quite a bit of delay at times, enough to make conversation awkward. As the setup at the far end was not completely trivial, I'm not 100% sure the problem was in my connection, but I'd like to test that. Are there any US numbers I can call to get an Asterisk-style echo test? Ideally, a California-based numnber, so I can try to call it from an ordinary PSTN phone here, and compare calling via VoIP, and see if there's an appreciable difference in the delay/quality. I don't anticipate using this for very long, so it doesn't necessarily need to be a free service. Failing that, does anyone have access to a US-based Asterisk server which would allow me to make connections to its echo test? Presumably, if I had this, I could rent a PSTN number from a US-based provider, and point it to the appropriate SIP/IAX address. I expect my total usage would be just a few minutes, though having the facility available for a few weeks would be helpful, to allow me to play around with various options. Again, I'd be willing to pay a modest amount for this. Thanks in advance for any suggestions! Best wishes, Nikhil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Thank you! [Was: Re: US-based echo test servers?]
Hi guys, Just wanted to say a big thank you to all the people who responded to this: I got far more help than I expected, and far more quickly than I would have thought possible. I'll be writing to a couple of you off-list, but didn't want to leave the list messages unanswered. I think, with these options, I should have ample opportunity to test things properly. BTW, Igor and Atis: sorry, I certainly didn't intend to start any sort of an argument, however short! For what it's worth, I feel you've both been generous, and I very much appreciate that. Best regards, Nikhil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US-based echo test servers?
I find smokeping (http://oss.oetiker.ch/smokeping/) to be very handy as a longer-term monitor of ping times. I have a section devoted to SIP peers on my home machine (see http://home.bod.org/smokeping/, click on the 'SIP Peers' graph). Darren raises a good point, though - those numbers are for the SIP hosts, not the media gateways. The numbers here are for a linux box on a fast, low-latency residential wireless connection, located in San Jose, CA. You may recognize a couple of the hosts as UK service providers (I'm an English ex-pat). -- Paul Darren Sessions wrote: Another thing you may want to do is try a simple ping test to the far end host. While this may not always be a reliable way to test lag given that the far end maybe just a proxy and your RTP may be terminating to another device, it still should give you a good idea what your lag times are at least on the signaling end of things. You could also do a traceroute to see how many hops your having to jump through as well. You could use a tool like ngrep to actually see the sip signaling and copy out the media gateway from the SDP if you really wanted to, and do a ping on that. I've done extensive work with international voip origination and termination, and typically I haven't had any problems unless it's going over satellite (lag) or there is a problem at the far end (usually pdd or quality issues). If things keep up, I'd even consider running top during a call to see what kind of utilization your local server is at just to make sure something isn't wrong there either. Hope this helps, - Darren _ [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 18, 2008, at 10:41 AM, Nikhil Nair wrote: Hi, I'm running a small Asterisk server in the UK, just for personal use. I've been experimenting with various VoIP providers for international calls to PSTN numbers, particularly to the US (often California). My results, to date, have been very variable indeed, so much so that I'm considering getting a suitable card and using the PSTN. I have found a VoIP provider with an excellent reputation, and it gives very good quality. However, I seem to get quite a bit of delay at times, enough to make conversation awkward. As the setup at the far end was not completely trivial, I'm not 100% sure the problem was in my connection, but I'd like to test that. Are there any US numbers I can call to get an Asterisk-style echo test? Ideally, a California-based numnber, so I can try to call it from an ordinary PSTN phone here, and compare calling via VoIP, and see if there's an appreciable difference in the delay/quality. I don't anticipate using this for very long, so it doesn't necessarily need to be a free service. Failing that, does anyone have access to a US-based Asterisk server which would allow me to make connections to its echo test? Presumably, if I had this, I could rent a PSTN number from a US-based provider, and point it to the appropriate SIP/IAX address. I expect my total usage would be just a few minutes, though having the facility available for a few weeks would be helpful, to allow me to play around with various options. Again, I'd be willing to pay a modest amount for this. Thanks in advance for any suggestions! Best wishes, Nikhil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] opening Doors with Asterisk!?
Hello all, i read a few articles online about the possibility to setup a buzzer door system to PBX using asterisk! currently my setup contains asterisk of course, and a sipura 3102.. what do i need to get such a feature done?! or should i ask if its possible?! _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opening Doors with Asterisk!?
You need a door phone or a switch to control the buzzer. I have used the 2N Entrycom and Helios line and they work very well. If you have an Astribank (Xorcom) you can use the output ports as switches. There are many brands of door phones you can choose from. You can connect them as regular extensions on the FXS port of the 3102 and control the door from there. On Mon, 2008-08-18 at 22:59 +0300, RoLaNd RoLaNd wrote: Hello all, i read a few articles online about the possibility to setup a buzzer door system to PBX using asterisk! currently my setup contains asterisk of course, and a sipura 3102.. what do i need to get such a feature done?! or should i ask if its possible?! __ Connect to the next generation of MSN Messenger Get it now! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opening Doors with Asterisk!?
Hello Ronald, I think there was a few threads going around not long ago from people who were trying to do this same thing. Try looking in the archives you might have some luck finding a good lead there. Regards, Igor H. RoLaNd RoLaNd wrote: Hello all, i read a few articles online about the possibility to setup a buzzer door system to PBX using asterisk! currently my setup contains asterisk of course, and a sipura 3102.. what do i need to get such a feature done?! or should i ask if its possible?! Connect to the next generation of MSN Messenger Get it now! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly my Asterisk Box Hanged up all calls
I've upgraded my version to 1.4.21.1 since last week and things seem to be fine. thanks, Daniel On Wed, Jul 23, 2008 at 11:27 AM, Chento Arohuanca [EMAIL PROTECTED]wrote: I´ll be upgrading my box this weekend and let you know the consequences. I´m new at the community and it would be good for me to know what was the problem with 1.4.17 Thanks for taking some time for me. Daniel On Wed, Jul 23, 2008 at 10:59 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Wed, Jul 23, 2008 at 10:44:12AM -0500, Tilghman Lesher wrote: On Wednesday 23 July 2008 10:15:18 Jay R. Ashworth wrote: On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote: On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote: My * version: 1.4.17 Please upgrade to 1.4.21.2. Just a suggestion, Tilghman: it might have been nice to add because it fixes your specific problem, so that we wouldn't assume because we don't want to talk to you if you rev is too old. :-) It probably fixes his specific problem, AND because I don't like diagnosing an issue that we've already solved and that he would have figured out, if he had bothered to try the latest release. 1.4.21.1 should have been fixed, as well, but at that point, I had just spent 3 hours working frantically to get two security advisories out the door, so that the community wouldn't be vulnerable to two critical issues, and suggesting that he try a version that was vulnerable would have been bad. Oh, sure. I'm just sayin... It's pretty clear to me that while you're playing in the NFL, he may not be. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange putconfig bahviour
On Monday 18 August 2008 12:02:15 Vadim Lebedev wrote: Value-01: 888,1,Noop(888) It may or may not be related, but those greater-than signs () in there look wrong, and may be causing some problems. Without them you can't get exten=888,1,Dial(888) if your remove '' you get exten=888,1,Dial(888) in extensions.conf which is BAD. Why is it bad? In all Asterisk config files, the '' after the '=' is superfluous for defining extensions, variables, etc. Try it. Having exten=123,1,... is perfectly valid and does not affect how Asterisk works in any appreciable way. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pollmailboxes
Philipp Kempgen wrote: I'm pretty sure there are IMAP servers with custom hooks (Dovecot?). Not exactly easy but doable. That's true. That's another thing that I would like to get implemented one of these days ... BTW: Does pollmailboxes _disable_ the event based notifications? UPGRADE.txt is not clear about that. Some setups might want to use a mix. Event-based with custom triggers and polling-based with pollfreq=3600 as a safety net. No, polling does not disable the event mechanism. The polling loop simply generates events when it detects changes. The rest of Asterisk still expects MWI to be handled in an event based fashion. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opening Doors with Asterisk!?
My existing setup is based on ITS Pantel device (http://its-tel.com/upload/Door_Release_5.6(1).pdf) that is connected to Linksys PAP2 device that is connected to Asterisk (SIP, of course). Works well, very flexible, quite cheap straight-forward setup. The only thing you need to put attention to is to set up the dial plan of the pap2 properly (to accept two digits dial, if I recall right). Carlos Chavez wrote: You need a door phone or a switch to control the buzzer. I have used the 2N Entrycom and Helios line and they work very well. If you have an Astribank (Xorcom) you can use the output ports as switches. There are many brands of door phones you can choose from. You can connect them as regular extensions on the FXS port of the 3102 and control the door from there. On Mon, 2008-08-18 at 22:59 +0300, RoLaNd RoLaNd wrote: Hello all, i read a few articles online about the possibility to setup a buzzer door system to PBX using asterisk! currently my setup contains asterisk of course, and a sipura 3102.. what do i need to get such a feature done?! or should i ask if its possible?! __ Connect to the next generation of MSN Messenger Get it now! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gnudialer runninig
Hi! I wanna know if here somebody has installed gnudialer ? I installed but i dont know how to run it Anybody has a cluee? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Get Windows Live and get whatever you need, wherever you are. Start here. http://www.windowslive.com/default.html?ocid=TXT_TAGLM_WL_Home_082008 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk AGI and php problem....
Chechk permissions *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* Date: Sat, 16 Aug 2008 13:20:18 -0400 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk AGI and php problem '/var/lib/asterisk/agi-bin/cid-to-acct.php': No such file or directory 2 == cid-to-acct.php: Failed to execute It is not complaining about the lack of /usr/bin/php, but about the fact that the file /var/lib/asterisk/agi-bin/cid-to-acct.php is nowhere to be found. Probably asking the obvious but... Did you place the file in the agi-bin folder ? Is it really named cid-to-acct.php ? Is it executable ? Does the user under which asterisk is running as the right to execute it ? hth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Your PC, mobile phone, and online services work together like never before. http://clk.atdmt.com/MRT/go/108587394/direct/01/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opening Doors with Asterisk!?
Viking, Valcom to name 2. They both use FXO ports. On Mon, Aug 18, 2008 at 3:59 PM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote: Hello all, i read a few articles online about the possibility to setup a buzzer door system to PBX using asterisk! currently my setup contains asterisk of course, and a sipura 3102.. what do i need to get such a feature done?! or should i ask if its possible?! Connect to the next generation of MSN Messenger Get it now! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Add Service Provider (was: Which is the correct GUI vor Asterisk 1.4 ?)
GUI used: current 2.0 branch out of SVN browser used: IE6 as well as FF 3 What is the proper way to add additional providers to the dropdown list? bkruseAll you have to do is click 'Add Service Provider'. I tried: Trunks - Service Provider - + New Service Provider. Then I get a popup that allows to select between Bandwith (SIP), Simple Signal (SIP), Voice Pulse (SIP and Voice Pulse (AIX). Hitting the ADD-Button only works if I previously marked one of the mentioned providers. Probably I haven´t explained my question sufficiently: How to add providers such as bluesip, sipgate, 11, ... ? Meaning a provider that is yet NOT listed in the list of, say available providers? I would expect that I would be asked a couple of questions (mainly those data that get´s written to providers.conf). Plus a possibility to upload a graphic for the provider. I tried to add a provider manually by changing the file providers.conf under /etc/asterisk but that didn´t work at all (the new provider doesn´t even get listed. There where even more providers already listed (IAXTEL, bandwidth, VoicePulse-sip, VoicePulse-iax, simplesignal, ngt) within providers.conf rather than those 4 available in the selection box of the Asterisk GUI. Strange However (which doesn´t imply that I´m happy with the workaround!) I was able to add SIP-providers via Trunks - VOIP Trunks - + New SIP/IAX Trunk. But that gave a new problem (which I will describe in a separate message) Best regards, Klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 44
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], altrimenti vi risponderò al mio rientro. Dimitri Osler I will be on vacation until Tuesday 19th of August with limited access to voice and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on my return. Dimitri Osler ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
